sip isup

Tags: CME cucm Cisco UnifiedThere is a logical association between the commands and must be configured in a sequential order.1. Configure voice encoding and SIP address formatVoice class URI 2 sipHost IPv4:address of CUCMVoice Class codec 1Codec Preference 1 G729R8Codec Preference 2 G711ulawCodec Preference 3 G711alawCodec Preference 4 g722-642. Configure VoIP Service Specific parametersVoice service VoIPIP Address Trusted ListIPv4 The address of the

SIP Reply Message Status codeand function Type Status Code status descriptionTemporary response (1XX) trying is in process180 ringing ringing181 Call being forwarder calls are forwardLine queue181* Session Progress Sessions Session succeeded (2XX) OK session succeeded Redirect (3XX) multiple multiple selectionMoved Permanently permanent move302 moved temporaily temporary movement305 Use Proxy User agent380 Alternative Service Alternative services Req

HTTP Authentication SIP provides a stateless, trial-and-error mechanism for the authentication system. This mechanism is based on HTTP authentication. At any time, the proxy server or UA receives a request (except in section 22.1), which attempts to check the identity confirmation provided by the request initiator. When the sender confirms the identity, the request recipient should confirm whether the user has been authenticated. In this document, it

SIP is one of the most important protocols in VoIP services. For this protocol, we have discussed some basic content related to it in some previous articles. We will not go into detail here. The focus today is to explain the knowledge about the SIP routing mechanism. In general, the SIP routing mechanism includes two scenarios: 1. Request message routing 2. Respo

Developed for SIP/IMS video clients, supports access to sip Softswitch, IMS core network, andVoice, video, and instant messaging functions. The video formats support h263, h264, and MPEG4 soft encoding solutions. The hardware coding/decoding interface is provided for interconnection and servers. If you are interested, contact me. Implements standard-based (SIP

At the request of the dynaguy brothers, I have attached a relatively complete experiment to colleagues who are still exploring: (There are some problems with the 2.0beta experiment, so I will demonstrate it with the most stable 1.2.3) In this test, we did not discuss the issue of using the SIP Trunk directly to connect other sip servers without board installation to enable the

In the process of building PYQT I met a very disgusting problem, in the installation of SIP after compiling the source of the installation process has been prompted me: Operation not permitted , I even reinstall the system is useless, and finally through the data to solve the problem. Installing SIPDownload the SIP source package after extracting it into its directory:python configure.pysudo makesudo m

SIP Redirect The SIP Redirect feature allows the IMG to respond to the 3xx class of SIP messages returned from a Redirect server. The 3XX responses provide information about a user ' s new location, or alternative services that is able to satisfy the Call. This feature was based on RFCs 3261 section 8.1.3.4 and RFC 2543. In a

SIP response code The response code is included, and the HTTP/1.1 response code is extended. Not all HTTP/1.1 Response codes are properly applied, but only pointed out in the discount. Other HTTP/1.1 Response codes should not be used. In addition, SIP also defines a new response code series, 6xx. 1. Temporary response 1xxA temporary response, that is, a message response, indicates that the other server is

The integration of unified communication involves software, hardware, mobile devices, and fixed devices. The standards and protocols are different. Therefore, the integration is difficult. In terms of IP phones and desktop applications, there are mainly SIP and H.323 Protocols; mobile videos are H.324M; Monitoring also includes H.323, SIP and proprietary protocols as well as MEPG2, MEPG4, and H. 264 and ot

Release date:Updated on: Affected Systems:Yealink Yealink SIP-T20P IP Phone Description:--------------------------------------------------------------------------------Bugtraq id: 57029Yealink SIP-T20P is an IP Phone.YeaLink IP Phone SIP-TxxP The vulnerability is described as follows:1) The default username ("user") and password ("user") can access the hidden pa

1 Description This article describes in detail the P2P SIP telephone process based on the STUN series protocol, which involves the interaction of SIP signaling, the principles of P2P, and Protocol interaction of STUN, TURN, and ICE. The interaction between service units mentioned in this article uses UDP, which does not involve TCP holes and other TCP-related operations. This document assumes that neither p

Both SIP and XMPP are application-layer protocols that are used primarily to send voice and instant messaging over the Internet im,rfc3521 defines the sip,rfc3920 definition of XMPP. XMPP comes from instant messaging systems, SIP-like voice and video communications.XMPP protocol is mainly responsible for the exchange of data,

Is the call flowchart of Asterisk: We use the call process of SIP as an example to describe the call process of other channels. The call process (incoming) is as follows: Do_monitor-> sipsock_read-> handle_request-> handle_request_invite-> sip_new/ast_pbx_start-> pbx_thread->__ ast_pbx_run -> Ast_spawn_extension-> pbx_extension_helper-> pbx_exec-> execute dialplan When the chan_sip module is loaded, an independent listening thread do_monitor is starte

Section 3 redirect servers In some frameworks, relying on the proxy server can reduce the load on the proxy server, which is beneficial for forwarding requests and enhancing signals. Redirection allows the server to send route information to the client through the response to the request, so it frees itself from the subsequent message loop of the transaction, at the same time, it can continue to accurately locate the request target. When the request's original sender receives a redirection, It r

Chapter 9 dialogue A key concept for user proxy is dialog. A dialog indicates a point-to-point sip connection between two user proxies at some time. The dialog ensures that messages between user proxies are ordered and correctly routed. A dialog indicates the context of a SIP message. Rfc3261 the UA processing discussed in section 8th is irrelevant to the method. This chapter discusses how to construct a di

Haha, if you have never touched network programming, don't look down. Give a definition first: SIP (Session Initial Protocol) is a signaling protocol that is used to set up, modify and terminate sessions, like Internet phone calland multimedia conferences between two participant ants. For Translation: SIP is a signaling protocol used to establish, modify, and terminate sessions, such as network call

1xx = notification response 100 trying 180 dialing in progress 181 being transferred 182 queuing 183 call progress 2XX = successful response 200 OK 202 accepted: used for referral 3xx = Transfer Response Over 300 options 301 permanent migration 302 temporarily migrated 305 use Proxy Server 380 alternative services 4xx = call failed 400 improper call 401 unauthorized: only for use by the Registry. The proxy server should use the proxy server for authorization 407 402 payment

Use FireBreath to develop real-time playback interfaces (Yate + SIP + FFMPEG + SDL) and firebreathyate At that time, such a blog post was really needed to guide this function module. Unfortunately, FireBreath has very little information on the Internet and is not very familiar with C ++, so we tried and explored it all the way. Fortunately, we have implemented this module, and now we have recorded it. First of all, our Yate

Tags: style blog http OS strong ar art Div log The SIP protocol is used in the national standard of the security video system. This document describes and develops a set of SIP protocol components. The exosip2 and osip2 libraries are generally used when developing such systems. This is an open-source SIP protocol stack library. The actual requirements cannot be